similar to: Normalize Voicemail Volume?

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on

We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know.

When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more

Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this

In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay

I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at AT&T, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when

Some simple refactoring of virt-resize. I originally had the idea that we could turn virt-resize into a library (cf. virt-customize) and use it from virt-builder, but I now don't think that would make any meaningful difference. In particular we'd still have to open the handle the same number of times. These two patches are left over from my work on that. Rich.

Hi all, Please a moment to look my source code, this is very similar to example of the documentation. I added FIXED_POINT flag. My source code compile and build but not decode correctly (the error may be in the while). I work with eVC 4.0 and pocket pc 2003. Someone know what is the error? Thanks. Rodrigo. #include "speex.h" #define FRAME_SIZE 160 void CPlayerDlg::OnButton3() {

SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

Parse the output disk as URI, and use all its attributes just like it is done for the input disk. The only change is that the fsync of the output disk is limited now for local URIs only, since it will not work with remote protocols. --- resize/resize.ml | 43 +++++++++++++++++++++++++++++++------------ resize/virt-resize.pod | 6 +++--- 2 files changed, 34 insertions(+), 15 deletions(-)

Previous patch contained a typo. Changed output_format -> output_filename. Rich.

On Thu, 2003-04-17 at 07:48, Kaveh Goudarzi wrote: > Hi, > > I had a similar question ... is the endian-ness of the encoded > speex file, system dependent? or is it always little endian? If it's > always little endian (like the header seems to be) then big endian > machines (or java) will need to map everything to bigendian before > decoding ... > I have spent some

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

Hello Speex, I think I've noticed a bug in version 1.1.6. In the source file "speexdec.c", lines 558 to 560 say wav_format = strlen(outFile)>=4 && ( strcmp(outFile+strlen(outFile)-4,".wav")==0 || strcmp(inFile+strlen(inFile)-4,".WAV")==0); . I think each

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many